Sorry I was wrong saying in a previous post that HDCD is 24 bits. HDCD is actually encoded on 20 bits but
the encoding scheme makes it compatible with 16 bits CD players.
So it will play at 16 bits on a normal CD player and at 20 bits on an HDCD compatible player.

Black magic and witcvhcraft.
Where's your proof - double blind listening tests or other validated test methodology.
The sort of 'analysis' you make above is similar to the high end hi-fi media.
It may be true, but you demonstrate no proof. The placebo effect is powerful. If I buy a new SACD machine, drop my new $big SCAD in the tray, smoothly glide it into the soft lit front panel and press play - my god its going to sound good!

So in this case we have a negative placebo effect?

He already owns a Rega Planet 2000 CD player priced at $900. This is truly a high-end audiophile unit. With such a unit in his system, the other units are probably sweet too.

The SACD player is a cheap low-end unit at $200 that quote: "sounds awful playing normal cds." At this price I would be surprised if it would even play music.

So we have three "results":

- A high-end CD player with an excellent 16bit DAC that sounds nice.
- A low-end SACD player with cheapest 24bit DAC that sounds awful when playing normal cds.
- A low-end SACD player with cheapest 24bit DAC that sounds nicer than the high end unit (if not nicer, then just as nice or maybe even close to jut as nice).

Double blind tests are nice. But in this case not relevant, as we are changing two factors: Player and media. Before performing the tests, we need some music in CD and SACD format mastered by 3rd party. I would like four different types:

A: Very subtle differences
B: Major improvements -- like comparing DVD-Video with VHS
C: The most interesting result. If the SACD comes even close to the CD, then there is hope for new music yet.
D: Time is money, so this is probably the most relevant test

I would go further and say that there is no chance of either of these formats having any kind of mass market. However, their development was not driven by customer's desires for multi-channel remixes of their favorite CDs (or their disgust with that wretched 16/44 sound). Their development was driven by the major label's demands for a secure format and their inability to make unrippable CDs. And I suspect that as soon as most new CD players will handle SACD and/or most DVD players will do DVD-Audio, then the phase-out of the CD will begin.

...assuming that the market puts up with that...

No new format has any chance for mass market before all hardware manufacturers and the label companies agree on a standard and work together.

History has shown that quality matters very little. VHS won over BetaMax, Dolby Digital is standard sound format in DVS-Video -- DTS is better... I own many DVDs with poor image quality... some of them worse than the VHS equilvalent.

Right now Sony is selling expensive SACD players to the few people who just "need" the newest technology at any price. When this phase is completed, we'll see what happens.

It isn't hard to convince the public to change -- especially not if both industries work togetger. Just keep releasing crap dymanic compressed CDs and good SACDs/DVD-As.

In my opinion, the sonic advantage of DVD-A and SACD is pure marketing hype, in an attempt to renovate part of the established base of CD players, also avoiding the possibility of making copies in these new formats.

It is also my opinion that standard redbook cd properly implemented (easy today) is transparent , I mean, beyond the capabilities of human hearing. All perceived differences in the new formats are due to a carefully done mastering.

I've read that JVC's XRCD standard redbook cd's, which are also carefully mastered, sound as good as SACD discs.

It would be interesting to know, as Pio pointed, at the recording/playing chain of these formats, how many of the microphones and processing devices (at the side of the recording) and speakers (at the side of the reproduction) are capable of recording, transmitting, or reproducing signals above 20-21 KHz and taking advantage of the 24 bits of dynamic range.

Besides, I've not read of any properly implemented blind test that showed perceived differences between those formats.

So, I find it very unlikely that the mass market is going to adopt a new system that offers no advantages (apart from multichannel sound), and is not copiable. In case of DVD and CD, they offer obvious superior quality and ease of use over VHS and vinyl.

If you have a 24/96 sound card, you can do you own 24/96 versus 16/44.1 tests with the material provided at:

It is physically *impossible* to record at 24bits w/o noise. There is always (thermal) noise. (To prevent this try to go next to 0°K...but then don't forget to take care of other maybe unwanted effects. ) But by this you'll have "natural" dither.

@Annuka

DTS only sounds better because of the bitrate. Dolby/DVD consortium was stupid enough to limit DD to 448kbps... I wonder how DTS would sound at that bitrate.

As Pio2001 explained, for SACD and DVD-A it is easier to sound better because of less quality DAC needed to produce good sound. But I have some doubt that the DA->AD->DA (player to amp to speaker) will do good for SACD/DVD-A...Yuck! (Thinking further, how many bits do you probably loose at the AD step again due to thermal noise?) So to assert SACD's pure sound you must connect it to an analogue amp, as all newer receivers work digital, so in the end you are listening to a SACD on-the-fly converted to DVD-A...
I am pretty sure with identical mastering an upsampled CD won't be distinguishable from DVD-A (if you limit DVD-A to 16bit; to get more bits for CD just encode to DTS or DD). SACD may sound different (but maybe not better) as it is purely based on dither.

I dunno how it is in the ultra high-end section, but usually digital amps are used in the the price regions I know. SACD can *never* sound better then DVD-A on such an amp due to the medium/technique! As I stated above the amp processes the signal digitally (mid-range: 48kHz at 16 or 24bit to high end 192khz at 24bit) the SACD will be converted DVD-A like, so if we can assume that it could be possible to connect DVD-A digitally the DA/AD steps will go away and we have a pure pcm transfer, DVD-A will sound more original than SACD.

If SACD does sound better on a digital amp it is due to mastering. I don't think that that DA/AD will improve sound...

So for me SACD makes absolutely *no* sense, and even sense of DVD-A is doubtful.

I dunno how it is in the ultra high-end section, but usually digital amps are used in the the price regions I know. SACD can *never* sound better then DVD-A on such an amp due to the medium/technique! As I stated above the amp processes the signal digitally (mid-range: 48kHz at 16 or 24bit to high end 192khz at 24bit) the SACD will be converted DVD-A like, so if we can assume that it could be possible to connect DVD-A digitally the DA/AD steps will go away and we have a pure pcm transfer, DVD-A will sound more original than SACD.

If SACD does sound better on a digital amp it is due to mastering. I don't think that that DA/AD will improve sound...

So for me SACD makes absolutely *no* sense, and even sense of DVD-A is doubtful.

I've heard that only the REALLY expensive models has digital out. And we are talking $6,000 +.

HDCD being capable of pseudo 20-bit playback only affects the dynamic range of the music, not the apparent quality of sound. As Ethan Winer uncovered in a semi-official test, most people can't identify the difference between 16 and 24-bit audio, unless dithering has been applied to the 16 bit audio. 24-bit provides advantages during the recording stage because the volume or a recorded track can be modified over a greater range without dropping any bits. Most music though is compressed down to having a 14-16dB or lower average dynamic range during the mastering stage. This is considered "good" dynamic range but most modern music squashed that down to something like 6dB i the interests of making the CD "loud." 24-bit properly recorded and converted to 16-bit theoretically should sound identical to a full 24-bit recording played back at 24-bit. A 24-bit recording would probably retain all the musical detail better than 16-bit would if your CD player has a digital volume control and you turned it down there rather than adjusting the analog volume of the amplifier.

KikeG mentioned that he didn't see the advantages of high sample rates if recording and playback equipment generally can't touch frequencies above 20Khz. The issue is really technical rather than purely being about the reproduction of nyquist frequencies. Part of it deals with the acuracy of reproducing given frequencies. At 44.1 you get 2.6 samples per 17Khz wav, at 96Khz you get 5.6 samples, at 192 you get 11 samples. Higher frequencies also inherently reduce the effects of jitter error, again, bringing high-end CD player clock stability to the average user at 96Khz. Another issue is distortions at or near the nyquist frequency. 44.1 sampled material must filter those out because they are still in an audible range, 96Khz pushes those well out beyond the range of human hearing allowing the DAC to do away with the filtering process. The filtering process, if done poorly can affect the quality of audio, contributing to the increase in clarity many describe when switching to higher sample rates.

In all cases you're bringing high-end audio quality to the mass market. It's a similar situation to what happened with CD's. Analog turntables and tape decks produced subjectively better sound, on a really good system with a really good turntable with a very well maintained album. However, CD players have lower noise and better clarity than the turntables or tape decks that most people used, so for some it was keeping things on par (albeit with lower noise and great conveniences like not destroying your albums as you played them) and for everyone else it was a great improvement.

Now, the rest of it, and probably the biggest reason its being pushed is monetary reasons. Selling people something new, and selling them something that can properly be copy-protected. In theory I agree with copy protection, especially as the artist's right to protect their property, but as a user who keeps all his music on his HD, I disagree with anything that prevents me from easily doing that. I especially disagree with the workaround of downloading MP3's made someone elses determined quality settings. Not only would they be MP3's instead of my prefered MPC's but they'd likely be at 128CBR since that's what the industry likes to tout as "CD quality."

...The issue is really technical rather than purely being about the reproduction of nyquist frequencies. Part of it deals with the acuracy of reproducing given frequencies. At 44.1 you get 2.6 samples per 17Khz wav, at 96Khz you get 5.6 samples, at 192 you get 11 samples. Higher frequencies also inherently reduce the effects of jitter error...

I'm not sure if you badly worded this or if you are misunderstanding something.

You present the number of samples for a given freq, and go on to say that 'higher frequencies also...', which implies that the number of samples in itself is an argument, which it isn't (and that's exactly what nyquist is about).

QUOTE

As Ethan Winer uncovered in a semi-official test, most people can't identify the difference between 16 and 24-bit audio, unless dithering has been applied to the 16 bit audio.

This is wrong. Most people could not identify the difference between the 16 and 13 bit files. There were two who could distinguish between the 16-bit ones, and _those_ preferred the truncated file on _subjective_ grounds. (http://ff123.net/24bit/24bitanalysis.html)

So, there you have it, SACD and DVD-Audio crippleware. On top of it all, SACDs and DVD-Audio discs cost more than CDs ( US +). Tough to see a mass market for these as much as they are technically impressive. Remember that 24bits gives you a dynamic range of 144dB - not many audio systems have that kind of accuracy, and it really become pointless in a car/airplane or even a home with an air conditioner.

Rant over.

Dynamic range is calculated by

6.02 dB * bits + 1.76 dB + Gain

Gain is around 15 dB for a sampling frequency of 44 kHz, around 36 dB for 96 kHz,
48 dB for 192 kHz.
It uses the effect taht the ATQ is not linear but frequency dependend.

I'm not sure if you badly worded this or if you are misunderstanding something.

You present the number of samples for a given freq, and go on to say that 'higher frequencies also...', which implies that the number of samples in itself is an argument, which it isn't (and that's exactly what nyquist is about).

I did in fact mean to state that higher sampling rates reduce the effects of jitter since, like the filtering I mentioned, it pushes jitter out beyond the point where humans can reasonably detect it. Thanks for catching my miswording. More samples do mean greater accuracy and faithfulness to the original analog signal, I did mean that portion of my statement. Nyquist as I understand it is not about accuracy, but determining the cutoff that a given sampling rate can reasonable reproduce. IMO 88.1 or 96Khz are sufficient, 192Khz would theoretically be better for accuracy, but I believe it's a point of diminishing returns, and with 192 the returns would be extremely diminished.

QUOTE

Originally posted by Garf

This is wrong. Most people could not identify the difference between the 16 and 13 bit files. There were two who could distinguish between the 16-bit ones, and _those_ preferred the truncated file on _subjective_ grounds. (http://ff123.net/24bit/24bitanalysis.html)

-- GCP

My mistake, the results still fully support my argument. Only via modifications to the source, or during the resampling process can anyone really discern a difference between varying bit-depths. Bit-depth allows for greater dynamic range, but not any inherent gain in audio quality. Theoretically, if done without the need for dithering, a downsampled 16/96Khz recording would sound indistinguishable from an original 24/96Khz recording. Not only that, but it would reduce the storage space requirements.

Originally posted by gdougherty More samples do mean greater accuracy and faithfulness to the original analog signal, I did mean that portion of my statement. Nyquist as I understand it is not about accuracy, but determining the cutoff that a given sampling rate can reasonable reproduce. IMO 88.1 or 96Khz are sufficient, 192Khz would theoretically be better for accuracy, but I believe it's a point of diminishing returns, and with 192 the returns would be extremely diminished.

You have a very fundamental misunderstanding. At 44khz sampling rate, you can reproduce a 20Khz signal [b]exactly as good as if you were sampling at 192Khz. This is the very essence of the nyquist theory. There is no gain whatsoever in increasing the sampling rate above two times the highest frequency that you want to record/reproduce.

If this feels intuitively strange, read up on the theory of sinc functions.

QUOTE

My mistake, the results still fully support my argument.

I agree with this. I made the correction because as written your original statement suggests it's better not to dither to get better accuracy in reproduction, which is false. (Not dithering may introduce additional distortion which can make the music sound better subjectively in some cases, which is what likely caused the curious result)

Dithering is _good_. You don't need to dither, but you _want_ to do it. The reason why the result of the test is seemingly at odds with this is that the listeners were not comparing to the 24-bit original, but only comparing among the two 16-bit files alone. They had no idea of determining which one was more true to the original, so they had to make a completely subjective pick among them. In that case, it's possible they pick the more distorted file because it has more ' pizzaz ' or whatever.

You have a very fundamental misunderstanding. At 44khz sampling rate, you can reproduce a 20Khz signal [b]exactly as good as if you were sampling at 192Khz. This is the very essence of the nyquist theory. There is no gain whatsoever in increasing the sampling rate above two times the highest frequency that you want to record/reproduce.

If this feels intuitively strange, read up on the theory of sinc functions.

I agree with this. I made the correction because as written your original statement suggests it's better not to dither to get better accuracy in reproduction, which is false. (Not dithering may introduce additional distortion which can make the music sound better subjectively in some cases, which is what likely caused the curious result)

Dithering is _good_. You don't need to dither, but you _want_ to do it. The reason why the result of the test is seemingly at odds with this is that the listeners were not comparing to the 24-bit original, but only comparing among the two 16-bit files alone. They had no idea of determining which one was more true to the original, so they had to make a completely subjective pick among them. In that case, it's possible they pick the more distorted file because it has more ' pizzaz ' or whatever.

-- GCP

That does sound counterintuitive. I had assumed there was a point at which it didn't matter anymore, and I guess due to the extremely short duration of the signal it makes some amount of sense, but it still seems odd. I shall have to read up on this. So then really the advantages of 96Khz would primarily come from reduced jitter effects (if you believe in jitter effecting audio quality) and dropping the need to filter out distortions inherent to the digital process.

Yes, dithering is not bad, where I made a point to say things should be done without dithering I was intending more to say that the same dynamic range would easily fit within 16-bits in most cases and between the two, not adding any dither into the situation would be best for comparison's sake.

As to your other questions, higher sampling frequencies will not benefit us when it comes to jitter because the amount of jitter between like amounts of time will be the same. The error from 1 44,100th of a second to the next will be still x picoseconds, regardless of what the frequency derived for clocking purposes is.

There's little left for 96 kHz sample rates, in theory.
Now all I want is listening, no player is perfect, and I still wonder if cheap 96 kHz converters sound better than 44.1 kHz ones.

At least recording a vinyl in 44.1 and 96 kHz 16 bits on the Marian Marc 2 soundcard, I sweared that the 96 kHz captured incredibly much more of the "analog" vinyl sound than the 44.1 kHz... until all audible differences suddenly vanished under the ABX hammer :cry2:

Originally posted by Pio2001 At least recording a vinyl in 44.1 and 96 kHz 16 bits on the Marian Marc 2 soundcard, I sweared that the 96 kHz captured incredibly much more of the "analog" vinyl sound than the 44.1 kHz... until all audible differences suddenly vanished under the ABX hammer :cry2:

You know, statements like this are so universal that it makes me question the absolute validity of ABX testing (which some people around here treat almost like a religion) for determining what is and isn’t important in audio reproduction. While it obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect. For example, I have seen a study that showed that PC monitor flicker had a significant effect on people’s tested reaction time even when it was well below the threshold of being perceived directly. It doesn’t seem to me that unlikely that the auditory system might not have some similar characteristics.

The clash between the objectivist and subjectivist viewpoints are pretty obvious in this thread, and it also seems like there is very little constructive communication between the two camps. It could be that the truth lies somewhere in that sparsely populated middle ground.

Subjective analysis s very important for unearthing audio problems. Noone can argue that if you can't measure something or can't ABX it, then, ergo, it doesn't exist. The Romans could not measure that the Earth was a sphere, but we now know it is (except for some oddballs).

However, what has to be done, is to put the acid on those that claim to hear something for a certain reason. That is, they must do more than make the assertion - they must provide some reasoned proof, with whatever methodlogy they can. The testing must start with a null hypothesis, there must be a control, each variable should be accounted for, the method must be described, the results must be repeatable, etc, etc.

Byant, I'm sure you are not arguing that the scientific method is flawed, are you? (Don't read this as me implying that the double blind test = the scientic method.) There is no middle ground when it comes to the results. You can use all the intuition you like to design the experiment, but the final proof must be based on cold, hard logic.

--------------------

Ruse____________________________Don't let the uncertainty turn you around,Go out and make a joyful sound.

Originally posted by bryant obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect. For example, I have seen a study that showed that PC monitor flicker had a significant effect on people’s tested reaction time even when it was well below the threshold of being perceived directly. It doesn’t seem to me that unlikely that the auditory system might not have some similar characteristics.

Then, it could be ABX'ed too. Maybe not with short tests, but with longer tests. Same applies to auditory system. If you can hear, feel, find any difference in any way, then it could be perceived also in a blind test. The only purpose of blind testing is to remove listener's expectation effect or bias, and make possible find differences only related to the sound.

QUOTE

You know, statements like this are so universal that it makes me question the absolute validity of ABX testing

Or question the validity of those statements no matter how apparently universal they are. I mean, they are "quite" universal for most people, but not so universal for more technical audio educated people.

While it obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect.

As other people have already pointed out, if they have an effect, they can be ABX-ed. What ABX does, is measure if, in any way, a listener can notice a difference between two clips. This means that the listener doesn't have to perceive anything wrong with either individually, or perceive (in the common sense of the word), a problem. If in any way they can determine a difference, they will be successfull.

A simple example is ABX-ing a lowpass based on a felt lack of 'air' .

If in no way they can determine a difference, the clips _are_ 'identical' to them for any means and purpose.

As Ruse pointed out, ABX is not the only way to work. Any scientifically valid method is good. The reason why we always refer to ABX tests is that they are easy to perform and analysing the results is well-understood. For example, the large scale listening tests we perform don't rely on ABX to get results, but the analysis is significantly more complicated than that of an ABX test.

It is often more difficult to ABX a difference than to hear it.
One of the only people that managed to ABX a 16 bits dithered file vs a 16 bits truncated file (while some other heard a difference but couldn't ABX it) performed one try each day ! Making a difference 16 times in a row can be very difficult.

Originally posted by Pio2001 When an ABX is positive, it does prove that the difference is audible, but when it's negative, it doesn't prove that there is no difference, it proves that the difference is "quite" inaudible.

Originally posted by Pio2001 It is often more difficult to ABX a difference than to hear it.
One of the only people that managed to ABX a 16 bits dithered file vs a 16 bits truncated file (while some other heard a difference but couldn't ABX it) performed one try each day ! Making a difference 16 times in a row can be very difficult.

...but it's perfectly possible if you hear a difference. Your argument is an example.

I think Pio is right. If you get a positive result, then you've proved you can hear a difference. But not being able to get a positive result doesn't mean you can't hear a difference, it only means that you've not proved you can hear a diference reliabily, so far.